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Hi, Has anybody else noticed that MOH does not play on parked calls in 1.6.2.14? Or is it just my setup? MOH seems to work in every other respect (Call Held or in-Queue), but when a call is parked, the logs show MOH being started, but the parked party hears nothing. The verbose logs show the following. Any thoughts on whet to check next? Thanks, Steve ### Call comes in here and is answered

Hello Although not stictly a asterisk issue, any help would be apreciated. Firstly a few notes on the snom 360, which I have had on a test bed for the last week. Its a great phone, with a good user interface, both physically and its web based one. At its lastest firmware it does have a few quirks, with regards to the way it handles usernames and passwords on the physical interface. These have

Here is my setup: * is on a NAT'd subnet, but also has an externally routable IP address. I have a Snom360 that's external to this and behind NAT. The Snom360 can call other phones in * subnet (by their internal extension numbers) and voice is transmitted fine; however, when I attempt to check voicemail (or any * voice recordings for that matter) I can't hear them. The phone just

Has anyone got this annoying sidecar to illuminate when users are on the phone? In my function key settings I have: Context: Active Type: Extension Number: <sip:4000@serve.address.com;user=phone> (4000 is the extension I want to see/dial on the key). I can press the key and it will dial the extension, it just won't illuminate when the user is on the phone or on DND Since I have

Asterisk seems to be ignoring DTMF for features in Asterisk 1.4.0b3 My SNOM sends the dtmf-relay and Asterisk gets it because I can see it in the sip debug. However, once seen, Asterisk doesn't actually do anything about it. I've checked features and that seems fine. Is this a bug or something that I've screwed up? For the record, here's the features setting: asterisk*CLI>

Sorry if this is an obvious question, but I haven't seen an obvious answer on the wiki that I remember. Has anyone managed to make the record button on the snom 360 fire off the Monitor() application? I don't see a bounty, and googling for "snom 360 record button asterisk" returns tons of product specification pages. (Joy!) I don't see a bounty for it, and the only

Good day all I have asterisk on a box with one network card I have a 2 companies setup on the system. To keep all apart I binded a different ip to the interface,i,o,w eth0 192.168.0.254 and eth0:1 192.168.1.254 And in sip.conf I took the bind setting out So each company's phones are on a different ip range,and all worked well So we decide to pull the snom190 out and exchange it with a snom360

Hi. I'm stuck into an odd situation. Here's what happens: 4 Thomson ST2030S 2 Cisco 7912 3 Cisco 7940 2 AAstra 480i Asterisk 1.2.17 Diva 4BRI + chan_capi I've just upgraded yesterday from Asterisk 1.2.13 to 1.2.17. Until yesterday, everything was just fine with 1.2.13. Immediately after the upgrade, *all* the 7940 are no more able to make calls, just receive them, while 7912

We are looking to replace our existing Legacy PBX with Asterisk. Our receptionist currently has a light display for a certain extension when someone is on a call. When she needs to transfer she simply hits that button. Is it possible to use a snom360 + Sidecar to monitor 30 extensions and make transfers using the buttons? Does the Cisco 7960 expansion module work with asterisk? Thanks.

Hi, I have successfully installed libunicall and mfcr2 in Venezuelan variant. We are able to receive, make calls without any problems. When we place a call from a snom sip phone through chan_unicall and press a dtmf (either using info, rfc2833 or inband) Asterisk seems to get the order of playing dtmf but a garbage is heard at the other side. I have been trying to narrow down the problem on

We have a SNOM360 (ext 226) configured for redirection (away on annual leave) to another SNOM360 (ext 225), being tested from a SNOM320 (ext 227) which appears on the surface to be an easy adjustment. Was receiving the following message, "Got SIP response 302 "Moved Temporarily" back from" of which, I was able to understand and correct by adding the following to the

Some of you might already know that we are releasing a new phone, snom 360. To make the phone well-known and stable, we have made a soft phone version out of it and offer it for trial or private use for free (for more details, see the license conditions). There are only few limitations to the phone. First of all, the audio subsystem will work only work with an acceptable quality if you are using

I got an up2date Asterisk with SNOM360 as SIP and mISDN with 2 ISDN Cards, if i press in a call the "*" Asterisk, Asterisk destroys the call not, Asterisk lets him hang and do nothing, if i hangup, Asterisk tell me in the warnings-log that the bridging was not successfull ?! If have disabled the function to hangup in the features.conf, but the key is still available, can someone

Hi, I want to pick up a call with the snom's programmable buttons(snom190 -SIP 3.60x, snom360-SIP 4.1) with asterisk server (v 1.2.0), I tried with the option 'Destination' and when the incoming call arrive to another snom phone the button blinking. In this way I can only pick down it pressing the blinking button. The solution is call the *8 or parcking the call but my pbroblem

Anyone have a Snom they're happy with? How did you manage that? :) I have a system of: Asterisk 1.2.3 2 Wildcard TDM400P Rev I and E/F 1 Snom 360 + sidecar ~15 Sipura/Linsys SPA-841 ~15 Grandstream 101 Everything (currently) is on the same network, not a router to be seen between any two. Also everything, except the snom, is working sweetly. The main problem is ECHO.. awful echo and

Hello, I am having an issue whereby calls are being dropped randomly. I have an ISDN 30 E1 line going into a Wildcard TE220 (4th Gen). My Asterisk install is based on Trixbox 2.0. However, I have updated the source code to the following. The Asterisk release is asterisk-1.2.20. Zaptel release is zaptel-1.2.18. And libpri release is libpri-1.2.4. I have include an extract from the Asterisk log

We have a large install of 360's running rev 4.1 with zero problems. I did another, smaller install couple weeks ago with 40 360's running rev 5.3. In both cases, the install was identical, same Asterisk version, same dialplan, everything the same except the differences were: 1. Different firmware rev 2. Different physical LAN Guess what? On the smaller install, lockups and reboots.

> > Hi, Remove usecallingpres=yes busydetect=yes from your zapata.conf file. and the restart asterisk. Hopefully you will not faced drop call issues. Regards, Vidura Senadeera. Message: 3 > Date: Mon, 24 Sep 2007 12:29:40 +0100 > From: "Richard Young" <Richard.Young at intrintech.com> > Subject: [asterisk-users] Asterisk Dropping Calls > To:

Long story short... Snom's ... Retrieve button... works when MWI is *NOT* lit but does *NOT* work when it is lit. Any advice Useragent : snom360/6.5.2 Function: F_RETRIEVE [root@pbx ~]# asterisk -rx "show version" Asterisk 1.2.13 built by root @ pbx on a i686 running Linux on 2006-11-17 16:35:22 UTC [gateway] exten => 201,hint,SIP/201 exten =>

Hi, I got a weird behaviour in asterisk (original found in 1.8 but it is still the same in 11.15.0). I have three phones communicating via OpenSIPs with asterisk. Phone A dials 100 and asterisk calls SIP/phone-b. Phone B accepts the call. The User on Phone B places the call on hold, dials 200 and, while hearing the dial tone of ringing Phone C, places the handset on hook. Phone B sends a REFER,